audio/filter: remove some useless filters

All of these filters are considered not useful anymore by us. Some have
replacements in libavfilter (useable through af_lavfi).

af_center, af_extrastereo, af_karaoke, af_sinesuppress, af_sub,
af_surround, af_sweep: pretty simple and useless filters which probably
nobody ever wants.

af_ladspa: has a replacement in libavfilter.

af_hrtf: the algorithm doesn't work properly on most sources, and the
implementation was buggy and complicated. (The filter was inherited from
MPlayer; but even in mpv times we had to apply fixes that fixed major
issues with added noise.) There is a ladspa filter if you still want to
use it.

af_export: I'm not even sure what this is supposed to do. Possibly it
was meant for GUIs rendering audio visualizations, but it couldn't
really work well. For example, the size of the audio depended on the
samplerate (fixed number of samples only), and it couldn't retrieve the
complete audio, only fragments. If this is really needed for GUIs, mpv
should add native visualization, or a proper API for it.

1 files changed, 0 insertions, 670 deletions

diff --git a/audio/filter/af_hrtf.c b/audio/filter/af_hrtf.c
deleted file mode 100644
index 3c8a89c665..0000000000
--- a/audio/filter/af_hrtf.c
+++ /dev/null
@@ -1,670 +0,0 @@
-/*
- * Experimental audio filter that mixes 5.1 and 5.1 with matrix
- * encoded rear channels into headphone signal using FIR filtering
- * with HRTF.
- *
- * Author: ylai
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv.  If not, see <http://www.gnu.org/licenses/>.
- */
-
-//#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <inttypes.h>
-
-#include <math.h>
-#include <libavutil/common.h>
-
-#include "af.h"
-#include "dsp.h"
-
-/* HRTF filter coefficients and adjustable parameters */
-#include "af_hrtf.h"
-
-typedef struct af_hrtf_s {
-    /* Lengths */
-    int dlbuflen, hrflen, basslen;
-    /* L, C, R, Ls, Rs channels */
-    float *lf, *rf, *lr, *rr, *cf, *cr;
-    const float *cf_ir, *af_ir, *of_ir, *ar_ir, *or_ir, *cr_ir;
-    int cf_o, af_o, of_o, ar_o, or_o, cr_o;
-    /* Bass */
-    float *ba_l, *ba_r;
-    float *ba_ir;
-    /* Whether to matrix decode the rear center channel */
-    int matrix_mode;
-    /* How to decode the input:
-       0 = 5/5+1 channels
-       1 = 2 channels
-       2 = matrix encoded 2 channels */
-    int decode_mode;
-    /* Full wave rectified (FWR) amplitudes and gain used to steer the
-       active matrix decoding of front channels (variable names
-       lpr/lmr means Lt + Rt, Lt - Rt) */
-    float l_fwr, r_fwr, lpr_fwr, lmr_fwr;
-    float adapt_l_gain, adapt_r_gain, adapt_lpr_gain, adapt_lmr_gain;
-    /* Matrix input decoding require special FWR buffer, since the
-       decoding is done in place. */
-    float *fwrbuf_l, *fwrbuf_r, *fwrbuf_lr, *fwrbuf_rr;
-    /* Rear channel delay buffer for matrix decoding */
-    float *rear_dlbuf;
-    /* Full wave rectified amplitude and gain used to steer the active
-       matrix decoding of center rear channel */
-    float lr_fwr, rr_fwr, lrprr_fwr, lrmrr_fwr;
-    float adapt_lr_gain, adapt_rr_gain;
-    float adapt_lrprr_gain, adapt_lrmrr_gain;
-    /* Cyclic position on the ring buffer */
-    int cyc_pos;
-    int print_flag;
-    int mode;
-} af_hrtf_t;
-
-/* Convolution on a ring buffer
- *    nx:       length of the ring buffer
- *    nk:       length of the convolution kernel
- *    sx:       ring buffer
- *    sk:       convolution kernel
- *    offset:   offset on the ring buffer, can be
- */
-static float conv(const int nx, const int nk, const float *sx, const float *sk,
-                  const int offset)
-{
-    /* k = reminder of offset / nx */
-    int k = offset >= 0 ? offset % nx : nx + (offset % nx);
-
-    if(nk + k <= nx)
-        return af_filter_fir(nk, sx + k, sk);
-    else
-        return af_filter_fir(nk + k - nx, sx, sk + nx - k) +
-            af_filter_fir(nx - k, sx + k, sk);
-}
-
-/* Detect when the impulse response starts (significantly) */
-static int pulse_detect(const float *sx)
-{
-    /* nmax must be the reference impulse response length (128) minus
-       s->hrflen */
-    const int nmax = 128 - HRTFFILTLEN;
-    const float thresh = IRTHRESH;
-    int i;
-
-    for(i = 0; i < nmax; i++)
-        if(fabs(sx[i]) > thresh)
-            return i;
-    return 0;
-}
-
-/* Fuzzy matrix coefficient transfer function to "lock" the matrix on
-   a effectively passive mode if the gain is approximately 1 */
-static inline float passive_lock(float x)
-{
-   const float x1 = x - 1;
-   const float ax1s = fabs(x - 1) * (1.0 / MATAGCLOCK);
-
-   return x1 - x1 / (1 + ax1s * ax1s) + 1;
-}
-
-/* Unified active matrix decoder for 2 channel matrix encoded surround
-   sources */
-static inline void matrix_decode(short *in, const int k, const int il,
-                          const int ir, const int decode_rear,
-                          const int dlbuflen,
-                          float l_fwr, float r_fwr,
-                          float lpr_fwr, float lmr_fwr,
-                          float *adapt_l_gain, float *adapt_r_gain,
-                          float *adapt_lpr_gain, float *adapt_lmr_gain,
-                          float *lf, float *rf, float *lr,
-                          float *rr, float *cf)
-{
-   const int kr = (k + MATREARDELAY) % dlbuflen;
-   float l_gain = (l_fwr + r_fwr) /
-      (1 + l_fwr + l_fwr);
-   float r_gain = (l_fwr + r_fwr) /
-      (1 + r_fwr + r_fwr);
-   /* The 2nd axis has strong gain fluctuations, and therefore require
-      limits.  The factor corresponds to the 1 / amplification of (Lt
-      - Rt) when (Lt, Rt) is strongly correlated. (e.g. during
-      dialogues).  It should be bigger than -12 dB to prevent
-      distortion. */
-   float lmr_lim_fwr = lmr_fwr > M9_03DB * lpr_fwr ?
-      lmr_fwr : M9_03DB * lpr_fwr;
-   float lpr_gain = (lpr_fwr + lmr_lim_fwr) /
-      (1 + lpr_fwr + lpr_fwr);
-   float lmr_gain = (lpr_fwr + lmr_lim_fwr) /
-      (1 + lmr_lim_fwr + lmr_lim_fwr);
-   float lmr_unlim_gain = (lpr_fwr + lmr_fwr) /
-      (1 + lmr_fwr + lmr_fwr);
-   float lpr, lmr;
-   float l_agc, r_agc, lpr_agc, lmr_agc;
-   float f, d_gain, c_gain, c_agc_cfk;
-
-#if 0
-   static int counter = 0;
-   static FILE *fp_out;
-
-   if(counter == 0)
-      fp_out = fopen("af_hrtf.log", "w");
-   if(counter % 240 == 0)
-      fprintf(fp_out, "%g %g %g %g %g ", counter * (1.0 / 48000),
-              l_gain, r_gain, lpr_gain, lmr_gain);
-#endif
-
-   /*** AXIS NO. 1: (Lt, Rt) -> (C, Ls, Rs) ***/
-   /* AGC adaption */
-   d_gain = (fabs(l_gain - *adapt_l_gain) +
-             fabs(r_gain - *adapt_r_gain)) * 0.5;
-   f = d_gain * (1.0 / MATAGCTRIG);
-   f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
-   *adapt_l_gain = (1 - f) * *adapt_l_gain + f * l_gain;
-   *adapt_r_gain = (1 - f) * *adapt_r_gain + f * r_gain;
-   /* Matrix */
-   l_agc = in[il] * passive_lock(*adapt_l_gain);
-   r_agc = in[ir] * passive_lock(*adapt_r_gain);
-   cf[k] = (l_agc + r_agc) * M_SQRT1_2;
-   if(decode_rear) {
-      lr[kr] = rr[kr] = (l_agc - r_agc) * M_SQRT1_2;
-      /* Stereo rear channel is steered with the same AGC steering as
-         the decoding matrix. Note this requires a fast updating AGC
-         at the order of 20 ms (which is the case here). */
-      lr[kr] *= (l_fwr + l_fwr) /
-         (1 + l_fwr + r_fwr);
-      rr[kr] *= (r_fwr + r_fwr) /
-         (1 + l_fwr + r_fwr);
-   }
-
-   /*** AXIS NO. 2: (Lt + Rt, Lt - Rt) -> (L, R) ***/
-   lpr = (in[il] + in[ir]) * M_SQRT1_2;
-   lmr = (in[il] - in[ir]) * M_SQRT1_2;
-   /* AGC adaption */
-   d_gain = fabs(lmr_unlim_gain - *adapt_lmr_gain);
-   f = d_gain * (1.0 / MATAGCTRIG);
-   f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
-   *adapt_lpr_gain = (1 - f) * *adapt_lpr_gain + f * lpr_gain;
-   *adapt_lmr_gain = (1 - f) * *adapt_lmr_gain + f * lmr_gain;
-   /* Matrix */
-   lpr_agc = lpr * passive_lock(*adapt_lpr_gain);
-   lmr_agc = lmr * passive_lock(*adapt_lmr_gain);
-   lf[k] = (lpr_agc + lmr_agc) * M_SQRT1_2;
-   rf[k] = (lpr_agc - lmr_agc) * M_SQRT1_2;
-
-   /*** CENTER FRONT CANCELLATION ***/
-   /* A heuristic approach exploits that Lt + Rt gain contains the
-      information about Lt, Rt correlation.  This effectively reshapes
-      the front and rear "cones" to concentrate Lt + Rt to C and
-      introduce Lt - Rt in L, R. */
-   /* 0.67677 is the empirical lower bound for lpr_gain. */
-   c_gain = 8 * (*adapt_lpr_gain - 0.67677);
-   c_gain = c_gain > 0 ? c_gain : 0;
-   /* c_gain should not be too high, not even reaching full
-      cancellation (~ 0.50 - 0.55 at current AGC implementation), or
-      the center will s0und too narrow. */
-   c_gain = MATCOMPGAIN / (1 + c_gain * c_gain);
-   c_agc_cfk = c_gain * cf[k];
-   lf[k] -= c_agc_cfk;
-   rf[k] -= c_agc_cfk;
-   cf[k] += c_agc_cfk + c_agc_cfk;
-#if 0
-   if(counter % 240 == 0)
-      fprintf(fp_out, "%g %g %g %g %g\n",
-              *adapt_l_gain, *adapt_r_gain,
-              *adapt_lpr_gain, *adapt_lmr_gain,
-              c_gain);
-   counter++;
-#endif
-}
-
-static inline void update_ch(af_hrtf_t *s, short *in, const int k)
-{
-    const int fwr_pos = (k + FWRDURATION) % s->dlbuflen;
-    /* Update the full wave rectified total amplitude */
-    /* Input matrix decoder */
-    if(s->decode_mode == HRTF_MIX_MATRIX2CH) {
-       s->l_fwr += abs(in[0]) - fabs(s->fwrbuf_l[fwr_pos]);
-       s->r_fwr += abs(in[1]) - fabs(s->fwrbuf_r[fwr_pos]);
-       s->lpr_fwr += abs(in[0] + in[1]) -
-          fabs(s->fwrbuf_l[fwr_pos] + s->fwrbuf_r[fwr_pos]);
-       s->lmr_fwr += abs(in[0] - in[1]) -
-          fabs(s->fwrbuf_l[fwr_pos] - s->fwrbuf_r[fwr_pos]);
-    }
-    /* Rear matrix decoder */
-    if(s->matrix_mode) {
-       s->lr_fwr += abs(in[2]) - fabs(s->fwrbuf_lr[fwr_pos]);
-       s->rr_fwr += abs(in[3]) - fabs(s->fwrbuf_rr[fwr_pos]);
-       s->lrprr_fwr += abs(in[2] + in[3]) -
-          fabs(s->fwrbuf_lr[fwr_pos] + s->fwrbuf_rr[fwr_pos]);
-       s->lrmrr_fwr += abs(in[2] - in[3]) -
-          fabs(s->fwrbuf_lr[fwr_pos] - s->fwrbuf_rr[fwr_pos]);
-    }
-
-    switch (s->decode_mode) {
-    case HRTF_MIX_51:
-       /* 5/5+1 channel sources */
-       s->lf[k] = in[0];
-       s->cf[k] = in[4];
-       s->rf[k] = in[1];
-       s->fwrbuf_lr[k] = s->lr[k] = in[2];
-       s->fwrbuf_rr[k] = s->rr[k] = in[3];
-       break;
-    case HRTF_MIX_MATRIX2CH:
-       /* Matrix encoded 2 channel sources */
-       s->fwrbuf_l[k] = in[0];
-       s->fwrbuf_r[k] = in[1];
-       matrix_decode(in, k, 0, 1, 1, s->dlbuflen,
-                     s->l_fwr, s->r_fwr,
-                     s->lpr_fwr, s->lmr_fwr,
-                     &(s->adapt_l_gain), &(s->adapt_r_gain),
-                     &(s->adapt_lpr_gain), &(s->adapt_lmr_gain),
-                     s->lf, s->rf, s->lr, s->rr, s->cf);
-       break;
-    case HRTF_MIX_STEREO:
-       /* Stereo sources */
-       s->lf[k] = in[0];
-       s->rf[k] = in[1];
-       s->cf[k] = s->lr[k] = s->rr[k] = 0;
-       break;
-    }
-
-    /* We need to update the bass compensation delay line, too. */
-    // TODO: should this use lf/cf/rf etc. instead?
-    s->ba_l[k] = in[0];
-    s->ba_r[k] = in[1];
-    if (s->decode_mode == HRTF_MIX_51) {
-        s->ba_l[k] += in[4] + in[2];
-        s->ba_r[k] += in[4] + in[3];
-    }
-}
-
-static void clear_coeff(af_hrtf_t *s, float *c)
-{
-    memset(c, 0, s->dlbuflen * sizeof(float));
-}
-
-static void reset(af_hrtf_t *s)
-{
-    clear_coeff(s, s->lf);
-    clear_coeff(s, s->rf);
-    clear_coeff(s, s->lr);
-    clear_coeff(s, s->rr);
-    clear_coeff(s, s->cf);
-    clear_coeff(s, s->cr);
-    clear_coeff(s, s->ba_l);
-    clear_coeff(s, s->ba_r);
-    clear_coeff(s, s->fwrbuf_l);
-    clear_coeff(s, s->fwrbuf_r);
-    clear_coeff(s, s->fwrbuf_lr);
-    clear_coeff(s, s->fwrbuf_rr);
-}
-
-/* Initialization and runtime control */
-static int control(struct af_instance *af, int cmd, void* arg)
-{
-    af_hrtf_t *s = af->priv;
-    int test_output_res;
-
-    switch(cmd) {
-    case AF_CONTROL_REINIT:
-        reset(s);
-        af->data->rate = 48000;
-        mp_audio_set_channels_old(af->data, ((struct mp_audio*)arg)->nch);
-        if(af->data->nch == 2) {
-            /* 2 channel input */
-            if(s->decode_mode != HRTF_MIX_MATRIX2CH) {
-                /* Default behavior is stereo mixing. */
-                s->decode_mode = HRTF_MIX_STEREO;
-            }
-        } else if (af->data->nch < 5) {
-            mp_audio_set_channels_old(af->data, 5);
-        }
-        mp_audio_set_format(af->data, AF_FORMAT_S16);
-        test_output_res = af_test_output(af, (struct mp_audio*)arg);
-        // after testing input set the real output format
-        mp_audio_set_num_channels(af->data, 2);
-        s->print_flag = 1;
-        return test_output_res;
-    case AF_CONTROL_RESET:
-        reset(s);
-        return AF_OK;
-    }
-
-    return AF_UNKNOWN;
-}
-
-/* Deallocate memory */
-static void uninit(struct af_instance *af)
-{
-        af_hrtf_t *s = af->priv;
-
-        free(s->lf);
-        free(s->rf);
-        free(s->lr);
-        free(s->rr);
-        free(s->cf);
-        free(s->cr);
-        free(s->ba_l);
-        free(s->ba_r);
-        free(s->ba_ir);
-        free(s->fwrbuf_l);
-        free(s->fwrbuf_r);
-        free(s->fwrbuf_lr);
-        free(s->fwrbuf_rr);
-}
-
-/* Filter data through filter
-
-Two "tricks" are used to compensate the "color" of the KEMAR data:
-
-1. The KEMAR data is refiltered to ensure that the front L, R channels
-on the same side of the ear are equalized (especially in the high
-frequencies).
-
-2. A bass compensation is introduced to ensure that 0-200 Hz are not
-damped (without any real 3D acoustical image, however).
-*/
-static int filter(struct af_instance *af, struct mp_audio *data)
-{
-    af_hrtf_t *s = af->priv;
-
-    if (!data)
-        return 0;
-    struct mp_audio *outframe =
-        mp_audio_pool_get(af->out_pool, &af->fmt_out, data->samples);
-    if (!outframe) {
-        talloc_free(data);
-        return -1;
-    }
-    mp_audio_copy_attributes(outframe, data);
-
-    short *in = data->planes[0]; // Input audio data
-    short *out = outframe->planes[0]; // Output audio data
-    short *end = in + data->samples * data->nch; // Loop end
-    float common, left, right, diff, left_b, right_b;
-    const int dblen = s->dlbuflen, hlen = s->hrflen, blen = s->basslen;
-
-    if(s->print_flag) {
-        s->print_flag = 0;
-        switch (s->decode_mode) {
-        case HRTF_MIX_51:
-          MP_INFO(af, "Using HRTF to mix %s discrete surround into "
-                 "L, R channels\n", s->matrix_mode ? "5+1" : "5");
-          break;
-        case HRTF_MIX_STEREO:
-          MP_INFO(af, "Using HRTF to mix stereo into "
-                 "L, R channels\n");
-          break;
-        case HRTF_MIX_MATRIX2CH:
-          MP_INFO(af, "Using active matrix to decode 2 channel "
-                 "input, HRTF to mix %s matrix surround into "
-                 "L, R channels\n", "3/2");
-          break;
-        default:
-          MP_WARN(af, "bogus decode_mode: %d\n", s->decode_mode);
-          break;
-        }
-
-       if(s->matrix_mode)
-          MP_INFO(af, "Using active matrix to decode rear center "
-                 "channel\n");
-    }
-
-    /* MPlayer's 5 channel layout (notation for the variable):
-     *
-     * 0: L (LF), 1: R (RF), 2: Ls (LR), 3: Rs (RR), 4: C (CF), matrix
-     * encoded: Cs (CR)
-     *
-     * or: L = left, C = center, R = right, F = front, R = rear
-     *
-     * Filter notation:
-     *
-     *      CF
-     * OF        AF
-     *      Ear->
-     * OR        AR
-     *      CR
-     *
-     * or: C = center, A = same side, O = opposite, F = front, R = rear
-     */
-
-    while(in < end) {
-        const int k = s->cyc_pos;
-
-        update_ch(s, in, k);
-
-        /* Simulate a 7.5 ms -20 dB echo of the center channel in the
-           front channels (like reflection from a room wall) - a kind of
-           psycho-acoustically "cheating" to focus the center front
-           channel, which is normally hard to be perceived as front */
-        s->lf[k] += CFECHOAMPL * s->cf[(k + CFECHODELAY) % s->dlbuflen];
-        s->rf[k] += CFECHOAMPL * s->cf[(k + CFECHODELAY) % s->dlbuflen];
-
-        switch (s->decode_mode) {
-        case HRTF_MIX_51:
-        case HRTF_MIX_MATRIX2CH:
-           /* Mixer filter matrix */
-           common = conv(dblen, hlen, s->cf, s->cf_ir, k + s->cf_o);
-           if(s->matrix_mode) {
-              /* In matrix decoding mode, the rear channel gain must be
-                 renormalized, as there is an additional channel. */
-              matrix_decode(in, k, 2, 3, 0, s->dlbuflen,
-                            s->lr_fwr, s->rr_fwr,
-                            s->lrprr_fwr, s->lrmrr_fwr,
-                            &(s->adapt_lr_gain), &(s->adapt_rr_gain),
-                            &(s->adapt_lrprr_gain), &(s->adapt_lrmrr_gain),
-                            s->lr, s->rr, NULL, NULL, s->cr);
-              common +=
-                 conv(dblen, hlen, s->cr, s->cr_ir, k + s->cr_o) *
-                 M1_76DB;
-              left    =
-                 ( conv(dblen, hlen, s->lf, s->af_ir, k + s->af_o) +
-                   conv(dblen, hlen, s->rf, s->of_ir, k + s->of_o) +
-                   (conv(dblen, hlen, s->lr, s->ar_ir, k + s->ar_o) +
-                    conv(dblen, hlen, s->rr, s->or_ir, k + s->or_o)) *
-                   M1_76DB + common);
-              right   =
-                 ( conv(dblen, hlen, s->rf, s->af_ir, k + s->af_o) +
-                   conv(dblen, hlen, s->lf, s->of_ir, k + s->of_o) +
-                   (conv(dblen, hlen, s->rr, s->ar_ir, k + s->ar_o) +
-                    conv(dblen, hlen, s->lr, s->or_ir, k + s->or_o)) *
-                   M1_76DB + common);
-           } else {
-              left    =
-                 ( conv(dblen, hlen, s->lf, s->af_ir, k + s->af_o) +
-                   conv(dblen, hlen, s->rf, s->of_ir, k + s->of_o) +
-                   conv(dblen, hlen, s->lr, s->ar_ir, k + s->ar_o) +
-                   conv(dblen, hlen, s->rr, s->or_ir, k + s->or_o) +
-                   common);
-              right   =
-                 ( conv(dblen, hlen, s->rf, s->af_ir, k + s->af_o) +
-                   conv(dblen, hlen, s->lf, s->of_ir, k + s->of_o) +
-                   conv(dblen, hlen, s->rr, s->ar_ir, k + s->ar_o) +
-                   conv(dblen, hlen, s->lr, s->or_ir, k + s->or_o) +
-                   common);
-           }
-           break;
-        case HRTF_MIX_STEREO:
-           left    =
-              ( conv(dblen, hlen, s->lf, s->af_ir, k + s->af_o) +
-                conv(dblen, hlen, s->rf, s->of_ir, k + s->of_o));
-           right   =
-              ( conv(dblen, hlen, s->rf, s->af_ir, k + s->af_o) +
-                conv(dblen, hlen, s->lf, s->of_ir, k + s->of_o));
-           break;
-        default:
-            /* make gcc happy */
-            left = 0.0;
-            right = 0.0;
-            break;
-        }
-
-        /* Bass compensation for the lower frequency cut of the HRTF.  A
-           cross talk of the left and right channel is introduced to
-           match the directional characteristics of higher frequencies.
-           The bass will not have any real 3D perception, but that is
-           OK (note at 180 Hz, the wavelength is about 2 m, and any
-           spatial perception is impossible). */
-        left_b  = conv(dblen, blen, s->ba_l, s->ba_ir, k);
-        right_b = conv(dblen, blen, s->ba_r, s->ba_ir, k);
-        left  += (1 - BASSCROSS) * left_b  + BASSCROSS * right_b;
-        right += (1 - BASSCROSS) * right_b + BASSCROSS * left_b;
-        /* Also mix the LFE channel (if available) */
-        if(data->nch >= 6) {
-            left  += in[5] * M3_01DB;
-            right += in[5] * M3_01DB;
-        }
-
-        /* Amplitude renormalization. */
-        left  *= AMPLNORM;
-        right *= AMPLNORM;
-
-        switch (s->decode_mode) {
-        case HRTF_MIX_51:
-        case HRTF_MIX_STEREO:
-           /* "Cheating": linear stereo expansion to amplify the 3D
-              perception.  Note: Too much will destroy the acoustic space
-              and may even result in headaches. */
-           diff = STEXPAND2 * (left - right);
-           out[0] = av_clip_int16(left  + diff);
-           out[1] = av_clip_int16(right - diff);
-           break;
-        case HRTF_MIX_MATRIX2CH:
-           /* Do attempt any stereo expansion with matrix encoded
-              sources.  The L, R channels are already stereo expanded
-              by the steering, any further stereo expansion will sound
-              very unnatural. */
-           out[0] = av_clip_int16(left);
-           out[1] = av_clip_int16(right);
-           break;
-        }
-
-        /* Next sample... */
-        in = &in[data->nch];
-        out = &out[af->data->nch];
-        (s->cyc_pos)--;
-        if(s->cyc_pos < 0)
-            s->cyc_pos += dblen;
-    }
-
-    talloc_free(data);
-    af_add_output_frame(af, outframe);
-    return 0;
-}
-
-static int allocate(af_hrtf_t *s)
-{
-    if ((s->lf = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->rf = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->lr = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->rr = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->cf = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->cr = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->ba_l = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->ba_r = malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->fwrbuf_l =
-         malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->fwrbuf_r =
-         malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->fwrbuf_lr =
-         malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    if ((s->fwrbuf_rr =
-         malloc(s->dlbuflen * sizeof(float))) == NULL) return -1;
-    return 0;
-}
-
-/* Allocate memory and set function pointers */
-static int af_open(struct af_instance* af)
-{
-    int i;
-    af_hrtf_t *s;
-    float fc;
-
-    af->control = control;
-    af->uninit = uninit;
-    af->filter_frame = filter;
-
-    s = af->priv;
-
-    s->dlbuflen = DELAYBUFLEN;
-    s->hrflen = HRTFFILTLEN;
-    s->basslen = BASSFILTLEN;
-
-    s->cyc_pos = s->dlbuflen - 1;
-    /* With a full (two axis) steering matrix decoder, s->matrix_mode
-       should not be enabled lightly (it will also steer the Ls, Rs
-       channels). */
-    s->matrix_mode = 0;
-    s->decode_mode = HRTF_MIX_51;
-
-    switch (s->mode) {
-    case 0: /* Use matrix rear decoding. */
-        s->matrix_mode = 1;
-        break;
-    case 1: /* Input needs matrix decoding. */
-        s->decode_mode = HRTF_MIX_MATRIX2CH;
-        break;
-    case 2:
-        s->matrix_mode = 0;
-        break;
-    }
-
-    s->print_flag = 1;
-
-    if (allocate(s) != 0) {
-        MP_ERR(af, "Memory allocation error.\n");
-        return AF_ERROR;
-    }
-
-    for(i = 0; i < s->dlbuflen; i++)
-        s->lf[i] = s->rf[i] = s->lr[i] = s->rr[i] = s->cf[i] =
-            s->cr[i] = 0;
-
-    s->lr_fwr =
-        s->rr_fwr = 0;
-
-    s->cf_ir = cf_filt + (s->cf_o = pulse_detect(cf_filt));
-    s->af_ir = af_filt + (s->af_o = pulse_detect(af_filt));
-    s->of_ir = of_filt + (s->of_o = pulse_detect(of_filt));
-    s->ar_ir = ar_filt + (s->ar_o = pulse_detect(ar_filt));
-    s->or_ir = or_filt + (s->or_o = pulse_detect(or_filt));
-    s->cr_ir = cr_filt + (s->cr_o = pulse_detect(cr_filt));
-
-    if((s->ba_ir = malloc(s->basslen * sizeof(float))) == NULL) {
-        MP_ERR(af, "Memory allocation error.\n");
-        return AF_ERROR;
-    }
-    fc = 2.0 * BASSFILTFREQ / (float)af->data->rate;
-    if(af_filter_design_fir(s->basslen, s->ba_ir, &fc, LP | KAISER, 4 * M_PI) ==
-       -1) {
-        MP_ERR(af, "Unable to design low-pass "
-               "filter.\n");
-        return AF_ERROR;
-    }
-    for(i = 0; i < s->basslen; i++)
-        s->ba_ir[i] *= BASSGAIN;
-
-    return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_hrtf_t
-const struct af_info af_info_hrtf = {
-    .info = "HRTF Headphone",
-    .name = "hrtf",
-    .open = af_open,
-    .priv_size = sizeof(af_hrtf_t),
-    .options = (const struct m_option[]) {
-        OPT_CHOICE("mode", mode, 0, ({"m", 0}, {"s", 1}, {"0", 2})),
-        {0}
-    },
-};